Update a ton of audio code to use size_t more correctly and in general reduce

webrtc/common_audio/resampler/push_sinc_resampler_unittest.cc

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 40 matching lines @@
   void ResampleTest(bool int_format);
 
   int input_rate_;
   int output_rate_;
   double rms_error_;
   double low_freq_error_;
 };
 
 class ZeroSource : public SincResamplerCallback {
  public:
-  void Run(int frames, float* destination) {
+  void Run(size_t frames, float* destination) {
     std::memset(destination, 0, sizeof(float) * frames);
   }
 };
 
 void PushSincResamplerTest::ResampleBenchmarkTest(bool int_format) {
-  const int input_samples = input_rate_ / 100;
-  const int output_samples = output_rate_ / 100;
+  const size_t input_samples = static_cast<size_t>(input_rate_ / 100);
+  const size_t output_samples = static_cast<size_t>(output_rate_ / 100);
   const int kResampleIterations = 500000;
 
   // Source for data to be resampled.
   ZeroSource resampler_source;
 
   rtc::scoped_ptr<float[]> resampled_destination(new float[output_samples]);
   rtc::scoped_ptr<float[]> source(new float[input_samples]);
   rtc::scoped_ptr<int16_t[]> source_int(new int16_t[input_samples]);
   rtc::scoped_ptr<int16_t[]> destination_int(new int16_t[output_samples]);
 
   resampler_source.Run(input_samples, source.get());
-  for (int i = 0; i < input_samples; ++i) {
+  for (size_t i = 0; i < input_samples; ++i) {
     source_int[i] = static_cast<int16_t>(floor(32767 * source[i] + 0.5));
   }
 
   printf("Benchmarking %d iterations of %d Hz -> %d Hz:\n",
          kResampleIterations, input_rate_, output_rate_);
   const double io_ratio = input_rate_ / static_cast<double>(output_rate_);
   SincResampler sinc_resampler(io_ratio, SincResampler::kDefaultRequestSize,
                                &resampler_source);
   TickTime start = TickTime::Now();
   for (int i = 0; i < kResampleIterations; ++i) {
@@ skipping 36 matching lines @@
 
 TEST_P(PushSincResamplerTest, DISABLED_BenchmarkFloat) {
   ResampleBenchmarkTest(false);
 }
 
 // Tests resampling using a given input and output sample rate.
 void PushSincResamplerTest::ResampleTest(bool int_format) {
   // Make comparisons using one second of data.
   static const double kTestDurationSecs = 1;
   // 10 ms blocks.
-  const int kNumBlocks = kTestDurationSecs * 100;
-  const int input_block_size = input_rate_ / 100;
-  const int output_block_size = output_rate_ / 100;
-  const int input_samples = kTestDurationSecs * input_rate_;
-  const int output_samples = kTestDurationSecs * output_rate_;
+  const size_t kNumBlocks = static_cast<size_t>(kTestDurationSecs * 100);
+  const size_t input_block_size = static_cast<size_t>(input_rate_ / 100);
+  const size_t output_block_size = static_cast<size_t>(output_rate_ / 100);
+  const size_t input_samples =
+      static_cast<size_t>(kTestDurationSecs * input_rate_);
+  const size_t output_samples =
+      static_cast<size_t>(kTestDurationSecs * output_rate_);
 
   // Nyquist frequency for the input sampling rate.
   const double input_nyquist_freq = 0.5 * input_rate_;
 
   // Source for data to be resampled.
   SinusoidalLinearChirpSource resampler_source(
       input_rate_, input_samples, input_nyquist_freq, 0);
 
   PushSincResampler resampler(input_block_size, output_block_size);
 
   // TODO(dalecurtis): If we switch to AVX/SSE optimization, we'll need to
   // allocate these on 32-byte boundaries and ensure they're sized % 32 bytes.
   rtc::scoped_ptr<float[]> resampled_destination(new float[output_samples]);
   rtc::scoped_ptr<float[]> pure_destination(new float[output_samples]);
   rtc::scoped_ptr<float[]> source(new float[input_samples]);
   rtc::scoped_ptr<int16_t[]> source_int(new int16_t[input_block_size]);
   rtc::scoped_ptr<int16_t[]> destination_int(new int16_t[output_block_size]);
 
   // The sinc resampler has an implicit delay of approximately half the kernel
   // size at the input sample rate. By moving to a push model, this delay
   // becomes explicit and is managed by zero-stuffing in PushSincResampler. We
   // deal with it in the test by delaying the "pure" source to match. It must be
   // checked before the first call to Resample(), because ChunkSize() will
   // change afterwards.
-  const int output_delay_samples = output_block_size -
+  const size_t output_delay_samples = output_block_size -
       resampler.get_resampler_for_testing()->ChunkSize();
 
   // Generate resampled signal.
   // With the PushSincResampler, we produce the signal block-by-10ms-block
   // rather than in a single pass, to exercise how it will be used in WebRTC.
   resampler_source.Run(input_samples, source.get());
   if (int_format) {
-    for (int i = 0; i < kNumBlocks; ++i) {
+    for (size_t i = 0; i < kNumBlocks; ++i) {
       FloatToS16(&source[i * input_block_size], input_block_size,
                source_int.get());
       EXPECT_EQ(output_block_size,
                 resampler.Resample(source_int.get(),
                                    input_block_size,
                                    destination_int.get(),
                                    output_block_size));
       S16ToFloat(destination_int.get(), output_block_size,
                &resampled_destination[i * output_block_size]);
     }
   } else {
-    for (int i = 0; i < kNumBlocks; ++i) {
+    for (size_t i = 0; i < kNumBlocks; ++i) {
       EXPECT_EQ(
           output_block_size,
           resampler.Resample(&source[i * input_block_size],
                              input_block_size,
                              &resampled_destination[i * output_block_size],
                              output_block_size));
     }
   }
 
   // Generate pure signal.
   SinusoidalLinearChirpSource pure_source(
       output_rate_, output_samples, input_nyquist_freq, output_delay_samples);
   pure_source.Run(output_samples, pure_destination.get());
 
   // Range of the Nyquist frequency (0.5 * min(input rate, output_rate)) which
   // we refer to as low and high.
   static const double kLowFrequencyNyquistRange = 0.7;
   static const double kHighFrequencyNyquistRange = 0.9;
 
   // Calculate Root-Mean-Square-Error and maximum error for the resampling.
   double sum_of_squares = 0;
   double low_freq_max_error = 0;
   double high_freq_max_error = 0;
   int minimum_rate = std::min(input_rate_, output_rate_);
   double low_frequency_range = kLowFrequencyNyquistRange * 0.5 * minimum_rate;
   double high_frequency_range = kHighFrequencyNyquistRange * 0.5 * minimum_rate;
 
-  for (int i = 0; i < output_samples; ++i) {
+  for (size_t i = 0; i < output_samples; ++i) {
     double error = fabs(resampled_destination[i] - pure_destination[i]);
 
     if (pure_source.Frequency(i) < low_frequency_range) {
       if (error > low_freq_max_error)
         low_freq_max_error = error;
     } else if (pure_source.Frequency(i) < high_frequency_range) {
       if (error > high_freq_max_error)
         high_freq_max_error = error;
     }
     // TODO(dalecurtis): Sanity check frequencies > kHighFrequencyNyquistRange.
@@ skipping 99 matching lines @@
         // To 32 kHz
         ::testing::make_tuple(8000, 32000, kResamplingRMSError, -70.30),
         ::testing::make_tuple(16000, 32000, kResamplingRMSError, -75.51),
         ::testing::make_tuple(32000, 32000, kResamplingRMSError, -75.51),
         ::testing::make_tuple(44100, 32000, -16.44, -51.10),
         ::testing::make_tuple(48000, 32000, -16.90, -44.03),
         ::testing::make_tuple(96000, 32000, -19.61, -18.04),
         ::testing::make_tuple(192000, 32000, -21.02, -10.94)));
 
 }  // namespace webrtc